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update software package

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This commit is contained in:
JeanLeo 2020-01-09 17:03:07 +08:00
parent ba3c548d39
commit b6466eff57
9 changed files with 75 additions and 347 deletions
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12
audiofile-0.3.6-CVE-2015-7747.patch
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@ -1,12 +0,0 @@
diff -Nurb --strip-trailing-cr audiofile-0.3.6-orig/libaudiofile/modules/ModuleState.cpp audiofile-0.3.6/libaudiofile/modules/ModuleState.cpp
--- audiofile-0.3.6-orig/libaudiofile/modules/ModuleState.cpp 2013-03-06 06:30:03.000000000 +0100
+++ audiofile-0.3.6/libaudiofile/modules/ModuleState.cpp 2015-10-08 11:29:51.846082162 +0200
@@ -402,7 +402,7 @@
addModule(new Transform(outfc, in.pcm, out.pcm));
if (in.channelCount != out.channelCount)
- addModule(new ApplyChannelMatrix(infc, isReading,
+ addModule(new ApplyChannelMatrix(outfc, isReading,
in.channelCount, out.channelCount,
in.pcm.minClip, in.pcm.maxClip,
track->channelMatrix));

48
audiofile-0.3.6-left-shift-neg.patch
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@ -1,48 +0,0 @@
diff -Nurb --strip-trailing-cr audiofile-0.3.6-orig/libaudiofile/modules/SimpleModule.h audiofile-0.3.6/libaudiofile/modules/SimpleModule.h
--- audiofile-0.3.6-orig/libaudiofile/modules/SimpleModule.h 2013-03-06 06:30:03.000000000 +0100
+++ audiofile-0.3.6/libaudiofile/modules/SimpleModule.h 2016-02-03 21:19:43.065454454 +0100
@@ -123,7 +123,7 @@
typedef typename IntTypes<Format>::UnsignedType UnsignedType;
static const int kScaleBits = (Format + 1) * CHAR_BIT - 1;
- static const int kMinSignedValue = -1 << kScaleBits;
+ static const int kMinSignedValue = 0-(1U<<kScaleBits);
struct signedToUnsigned : public std::unary_function<SignedType, UnsignedType>
{
diff -Nurb --strip-trailing-cr audiofile-0.3.6-orig/test/FloatToInt.cpp audiofile-0.3.6/test/FloatToInt.cpp
--- audiofile-0.3.6-orig/test/FloatToInt.cpp 2013-02-11 18:23:26.000000000 +0100
+++ audiofile-0.3.6/test/FloatToInt.cpp 2016-02-03 21:21:14.714510229 +0100
@@ -115,7 +115,7 @@
EXPECT_EQ(readData[i], expectedData[i]);
}
-static const int32_t kMinInt24 = -1<<23;
+static const int32_t kMinInt24 = 0-(1U<<23);
static const int32_t kMaxInt24 = (1<<23) - 1;
TEST_F(FloatToIntTest, Int24)
diff -Nurb --strip-trailing-cr audiofile-0.3.6-orig/test/IntToFloat.cpp audiofile-0.3.6/test/IntToFloat.cpp
--- audiofile-0.3.6-orig/test/IntToFloat.cpp 2013-02-11 18:23:26.000000000 +0100
+++ audiofile-0.3.6/test/IntToFloat.cpp 2016-02-03 21:20:57.380445355 +0100
@@ -117,7 +117,7 @@
EXPECT_EQ(readData[i], expectedData[i]);
}
-static const int32_t kMinInt24 = -1<<23;
+static const int32_t kMinInt24 = 0-(1U<<23);
static const int32_t kMaxInt24 = (1<<23) - 1;
TEST_F(IntToFloatTest, Int24)
diff -Nurb --strip-trailing-cr audiofile-0.3.6-orig/test/Sign.cpp audiofile-0.3.6/test/Sign.cpp
--- audiofile-0.3.6-orig/test/Sign.cpp 2013-02-11 18:23:26.000000000 +0100
+++ audiofile-0.3.6/test/Sign.cpp 2016-02-03 21:20:38.742450826 +0100
@@ -116,7 +116,7 @@
EXPECT_EQ(readData[i], expectedData[i]);
}
-static const int32_t kMinInt24 = -1<<23;
+static const int32_t kMinInt24 = 0-(1U<<23);
static const int32_t kMaxInt24 = (1<<23) - 1;
static const uint32_t kMaxUInt24 = (1<<24) - 1;

52
audiofile-0.3.6-narrowing.patch
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@ -1,52 +0,0 @@
diff -Nur audiofile-0.3.6-orig/test/NeXT.cpp audiofile-0.3.6/test/NeXT.cpp
--- audiofile-0.3.6-orig/test/NeXT.cpp 2013-02-11 18:23:26.000000000 +0100
+++ audiofile-0.3.6/test/NeXT.cpp 2016-02-04 10:37:32.457140823 +0100
@@ -37,13 +37,13 @@
#include "TestUtilities.h"
-const char kDataUnspecifiedLength[] =
+const signed char kDataUnspecifiedLength[] =
{
'.', 's', 'n', 'd',
0, 0, 0, 24, // offset of 24 bytes
- 0xff, 0xff, 0xff, 0xff, // unspecified length
+ -1, -1, -1, -1, // unspecified length
0, 0, 0, 3, // 16-bit linear
- 0, 0, 172, 68, // 44100 Hz
+ 0, 0, -84, 68, // 44100 Hz (0xAC44)
0, 0, 0, 1, // 1 channel
0, 1,
0, 1,
@@ -57,13 +57,13 @@
0, 55
};
-const char kDataTruncated[] =
+const signed char kDataTruncated[] =
{
'.', 's', 'n', 'd',
0, 0, 0, 24, // offset of 24 bytes
0, 0, 0, 20, // length of 20 bytes
0, 0, 0, 3, // 16-bit linear
- 0, 0, 172, 68, // 44100 Hz
+ 0, 0, -84, 68, // 44100 Hz (0xAC44)
0, 0, 0, 1, // 1 channel
0, 1,
0, 1,
@@ -152,13 +152,13 @@
ASSERT_EQ(::unlink(testFileName.c_str()), 0);
}
-const char kDataZeroChannels[] =
+const signed char kDataZeroChannels[] =
{
'.', 's', 'n', 'd',
0, 0, 0, 24, // offset of 24 bytes
0, 0, 0, 2, // 2 bytes
0, 0, 0, 3, // 16-bit linear
- 0, 0, 172, 68, // 44100 Hz
+ 0, 0, -84, 68, // 44100 Hz (0xAC44)
0, 0, 0, 0, // 0 channels
0, 1
};

176
audiofile-0.3.6-pull42.patch
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@ -1,176 +0,0 @@
diff -Nur audiofile-0.3.6/libaudiofile/modules/BlockCodec.cpp audiofile-0.3.6-pull42/libaudiofile/modules/BlockCodec.cpp
--- audiofile-0.3.6/libaudiofile/modules/BlockCodec.cpp 2013-03-06 06:30:03.000000000 +0100
+++ audiofile-0.3.6-pull42/libaudiofile/modules/BlockCodec.cpp 2017-03-10 15:40:02.000000000 +0100
@@ -52,8 +52,9 @@
// Decompress into m_outChunk.
for (int i=0; i<blocksRead; i++)
{
- decodeBlock(static_cast<const uint8_t *>(m_inChunk->buffer) + i * m_bytesPerPacket,
- static_cast<int16_t *>(m_outChunk->buffer) + i * m_framesPerPacket * m_track->f.channelCount);
+ if (decodeBlock(static_cast<const uint8_t *>(m_inChunk->buffer) + i * m_bytesPerPacket,
+ static_cast<int16_t *>(m_outChunk->buffer) + i * m_framesPerPacket * m_track->f.channelCount)==0)
+ break;
framesRead += m_framesPerPacket;
}
diff -Nur audiofile-0.3.6/libaudiofile/modules/MSADPCM.cpp audiofile-0.3.6-pull42/libaudiofile/modules/MSADPCM.cpp
--- audiofile-0.3.6/libaudiofile/modules/MSADPCM.cpp 2013-03-06 06:30:03.000000000 +0100
+++ audiofile-0.3.6-pull42/libaudiofile/modules/MSADPCM.cpp 2017-03-10 15:40:02.000000000 +0100
@@ -101,24 +101,60 @@
768, 614, 512, 409, 307, 230, 230, 230
};
+int firstBitSet(int x)
+{
+ int position=0;
+ while (x!=0)
+ {
+ x>>=1;
+ ++position;
+ }
+ return position;
+}
+
+#ifndef __has_builtin
+#define __has_builtin(x) 0
+#endif
+
+bool multiplyCheckOverflow(int a, int b, int *result)
+{
+#if (defined __GNUC__ && __GNUC__ >= 5) || ( __clang__ && __has_builtin(__builtin_mul_overflow))
+ return __builtin_mul_overflow(a, b, result);
+#else
+ if (firstBitSet(a)+firstBitSet(b)>31) // int is signed, so we can't use 32 bits
+ return true;
+ *result = a * b;
+ return false;
+#endif
+}
+
+
// Compute a linear PCM value from the given differential coded value.
static int16_t decodeSample(ms_adpcm_state &state,
- uint8_t code, const int16_t *coefficient)
+ uint8_t code, const int16_t *coefficient, bool *ok=NULL)
{
int linearSample = (state.sample1 * coefficient[0] +
state.sample2 * coefficient[1]) >> 8;
+ int delta;
linearSample += ((code & 0x08) ? (code - 0x10) : code) * state.delta;
linearSample = clamp(linearSample, MIN_INT16, MAX_INT16);
- int delta = (state.delta * adaptationTable[code]) >> 8;
+ if (multiplyCheckOverflow(state.delta, adaptationTable[code], &delta))
+ {
+ if (ok) *ok=false;
+ _af_error(AF_BAD_COMPRESSION, "Error decoding sample");
+ return 0;
+ }
+ delta >>= 8;
if (delta < 16)
delta = 16;
state.delta = delta;
state.sample2 = state.sample1;
state.sample1 = linearSample;
+ if (ok) *ok=true;
return static_cast<int16_t>(linearSample);
}
@@ -212,13 +248,16 @@
{
uint8_t code;
int16_t newSample;
+ bool ok;
code = *encoded >> 4;
- newSample = decodeSample(*state[0], code, coefficient[0]);
+ newSample = decodeSample(*state[0], code, coefficient[0], &ok);
+ if (!ok) return 0;
*decoded++ = newSample;
code = *encoded & 0x0f;
- newSample = decodeSample(*state[1], code, coefficient[1]);
+ newSample = decodeSample(*state[1], code, coefficient[1], &ok);
+ if (!ok) return 0;
*decoded++ = newSample;
encoded++;
diff -Nur audiofile-0.3.6/libaudiofile/WAVE.cpp audiofile-0.3.6-pull42/libaudiofile/WAVE.cpp
--- audiofile-0.3.6/libaudiofile/WAVE.cpp 2013-03-06 06:30:03.000000000 +0100
+++ audiofile-0.3.6-pull42/libaudiofile/WAVE.cpp 2017-03-10 15:40:02.000000000 +0100
@@ -281,6 +281,12 @@
/* numCoefficients should be at least 7. */
assert(numCoefficients >= 7 && numCoefficients <= 255);
+ if (numCoefficients < 7 || numCoefficients > 255)
+ {
+ _af_error(AF_BAD_HEADER,
+ "Bad number of coefficients");
+ return AF_FAIL;
+ }
m_msadpcmNumCoefficients = numCoefficients;
@@ -834,6 +840,8 @@
}
TrackSetup *track = setup->getTrack();
+ if (!track)
+ return AF_NULL_FILESETUP;
if (track->f.isCompressed())
{
diff -Nur audiofile-0.3.6/sfcommands/sfconvert.c audiofile-0.3.6-pull42/sfcommands/sfconvert.c
--- audiofile-0.3.6/sfcommands/sfconvert.c 2013-03-06 06:30:03.000000000 +0100
+++ audiofile-0.3.6-pull42/sfcommands/sfconvert.c 2017-03-10 15:40:02.000000000 +0100
@@ -45,6 +45,33 @@
void usageerror (void);
bool copyaudiodata (AFfilehandle infile, AFfilehandle outfile, int trackid);
+int firstBitSet(int x)
+{
+ int position=0;
+ while (x!=0)
+ {
+ x>>=1;
+ ++position;
+ }
+ return position;
+}
+
+#ifndef __has_builtin
+#define __has_builtin(x) 0
+#endif
+
+bool multiplyCheckOverflow(int a, int b, int *result)
+{
+#if (defined __GNUC__ && __GNUC__ >= 5) || ( __clang__ && __has_builtin(__builtin_mul_overflow))
+ return __builtin_mul_overflow(a, b, result);
+#else
+ if (firstBitSet(a)+firstBitSet(b)>31) // int is signed, so we can't use 32 bits
+ return true;
+ *result = a * b;
+ return false;
+#endif
+}
+
int main (int argc, char **argv)
{
if (argc == 2)
@@ -323,8 +350,11 @@
{
int frameSize = afGetVirtualFrameSize(infile, trackid, 1);
- const int kBufferFrameCount = 65536;
- void *buffer = malloc(kBufferFrameCount * frameSize);
+ int kBufferFrameCount = 65536;
+ int bufferSize;
+ while (multiplyCheckOverflow(kBufferFrameCount, frameSize, &bufferSize))
+ kBufferFrameCount /= 2;
+ void *buffer = malloc(bufferSize);
AFframecount totalFrames = afGetFrameCount(infile, AF_DEFAULT_TRACK);
AFframecount totalFramesWritten = 0;

21
audiofile-0.3.6-pull43.patch
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@ -1,21 +0,0 @@
diff -Nur audiofile-0.3.6/libaudiofile/modules/IMA.cpp audiofile-0.3.6-pull43/libaudiofile/modules/IMA.cpp
--- audiofile-0.3.6/libaudiofile/modules/IMA.cpp 2013-03-06 06:30:03.000000000 +0100
+++ audiofile-0.3.6-pull43/libaudiofile/modules/IMA.cpp 2017-03-06 18:06:35.000000000 +0100
@@ -169,7 +169,7 @@
if (encoded[1] & 0x80)
m_adpcmState[c].previousValue -= 0x10000;
- m_adpcmState[c].index = encoded[2];
+ m_adpcmState[c].index = clamp(encoded[2], 0, 88);
*decoded++ = m_adpcmState[c].previousValue;
@@ -210,7 +210,7 @@
predictor -= 0x10000;
state.previousValue = clamp(predictor, MIN_INT16, MAX_INT16);
- state.index = encoded[1] & 0x7f;
+ state.index = clamp(encoded[1] & 0x7f, 0, 88);
encoded += 2;
for (int n=0; n<m_framesPerPacket; n+=2)

31
audiofile-0.3.6-pull44.patch
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@ -1,31 +0,0 @@
diff -Nur audiofile-0.3.6/libaudiofile/modules/BlockCodec.cpp audiofile-0.3.6-pull44/libaudiofile/modules/BlockCodec.cpp
--- audiofile-0.3.6/libaudiofile/modules/BlockCodec.cpp 2013-03-06 06:30:03.000000000 +0100
+++ audiofile-0.3.6-pull44/libaudiofile/modules/BlockCodec.cpp 2017-03-09 10:21:18.000000000 +0100
@@ -47,7 +47,7 @@
// Read the compressed data.
ssize_t bytesRead = read(m_inChunk->buffer, m_bytesPerPacket * blockCount);
- int blocksRead = bytesRead >= 0 ? bytesRead / m_bytesPerPacket : 0;
+ int blocksRead = (bytesRead >= 0 && m_bytesPerPacket > 0) ? bytesRead / m_bytesPerPacket : 0;
// Decompress into m_outChunk.
for (int i=0; i<blocksRead; i++)
diff -Nur audiofile-0.3.6/libaudiofile/WAVE.cpp audiofile-0.3.6-pull44/libaudiofile/WAVE.cpp
--- audiofile-0.3.6/libaudiofile/WAVE.cpp 2013-03-06 06:30:03.000000000 +0100
+++ audiofile-0.3.6-pull44/libaudiofile/WAVE.cpp 2017-03-09 10:21:18.000000000 +0100
@@ -326,6 +326,7 @@
{
_af_error(AF_BAD_NOT_IMPLEMENTED,
"IMA ADPCM compression supports only 4 bits per sample");
+ return AF_FAIL;
}
int bytesPerBlock = (samplesPerBlock + 14) / 8 * 4 * channelCount;
@@ -333,6 +334,7 @@
{
_af_error(AF_BAD_CODEC_CONFIG,
"Invalid samples per block for IMA ADPCM compression");
+ return AF_FAIL;
}
track->f.sampleWidth = 16;

13
audiofile-CVE-2015-7747.patch Normal file
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@ -0,0 +1,13 @@
diff --git a/libaudiofile/modules/ModuleState.cpp b/libaudiofile/modules/ModuleState.cpp
index f76c495..0c29d7a 100644
--- a/libaudiofile/modules/ModuleState.cpp
+++ b/libaudiofile/modules/ModuleState.cpp
@@ -402,7 +402,7 @@ status ModuleState::arrange(AFfilehandle file, Track *track)
addModule(new Transform(outfc, in.pcm, out.pcm));
if (in.channelCount != out.channelCount)
- addModule(new ApplyChannelMatrix(infc, isReading,
+ addModule(new ApplyChannelMatrix(outfc, isReading,
in.channelCount, out.channelCount,
in.pcm.minClip, in.pcm.maxClip,
track->channelMatrix));

53
audiofile-fix-gcc6-compile-error.patch Normal file
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@ -0,0 +1,53 @@
diff --git a/libaudiofile/modules/SimpleModule.h b/libaudiofile/modules/SimpleModule.h
index 03c6c69..8e453a7 100644
--- a/libaudiofile/modules/SimpleModule.h
+++ b/libaudiofile/modules/SimpleModule.h
@@ -123,7 +123,7 @@ struct signConverter
typedef typename IntTypes<Format>::UnsignedType UnsignedType;
static const int kScaleBits = (Format + 1) * CHAR_BIT - 1;
- static const int kMinSignedValue = -1 << kScaleBits;
+ static const int kMinSignedValue = -1U << kScaleBits;
struct signedToUnsigned : public std::unary_function<SignedType, UnsignedType>
{
diff --git a/test/FloatToInt.cpp b/test/FloatToInt.cpp
index 0d179a8..624be25 100644
--- a/test/FloatToInt.cpp
+++ b/test/FloatToInt.cpp
@@ -115,7 +115,7 @@ TEST_F(FloatToIntTest, Int16)
EXPECT_EQ(readData[i], expectedData[i]);
}
-static const int32_t kMinInt24 = -1<<23;
+static const int32_t kMinInt24 = -1U<<23;
static const int32_t kMaxInt24 = (1<<23) - 1;
TEST_F(FloatToIntTest, Int24)
diff --git a/test/IntToFloat.cpp b/test/IntToFloat.cpp
index b716635..b619a2c 100644
--- a/test/IntToFloat.cpp
+++ b/test/IntToFloat.cpp
@@ -117,7 +117,7 @@ TEST_F(IntToFloatTest, Int16)
EXPECT_EQ(readData[i], expectedData[i]);
}
-static const int32_t kMinInt24 = -1<<23;
+static const int32_t kMinInt24 = -1U<<23;
static const int32_t kMaxInt24 = (1<<23) - 1;
TEST_F(IntToFloatTest, Int24)
diff --git a/test/Sign.cpp b/test/Sign.cpp
index 7275399..b3609ca 100644
--- a/test/Sign.cpp
+++ b/test/Sign.cpp
@@ -116,7 +116,7 @@ TEST_F(SignConversionTest, Int16)
EXPECT_EQ(readData[i], expectedData[i]);
}
-static const int32_t kMinInt24 = -1<<23;
+static const int32_t kMinInt24 = -1U<<23;
static const int32_t kMaxInt24 = (1<<23) - 1;
static const uint32_t kMaxUInt24 = (1<<24) - 1;

16
audiofile.spec
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@ -1,16 +1,12 @@
Name: audiofile Name: audiofile
Version: 0.3.6 Version: 0.3.6
Release: 22 Release: 23
Summary: Library for reading and writing audio files in many common formats Summary: Library for reading and writing audio files in many common formats
License: LGPLv2+ and GPLv2+ License: LGPLv2+ and GPLv2+
URL: http://audiofile.68k.org/ URL: http://audiofile.68k.org/
Source0: http://audiofile.68k.org/%{name}-%{version}.tar.gz Source0: http://audiofile.68k.org/%{name}-%{version}.tar.gz
Patch0: audiofile-0.3.6-CVE-2015-7747.patch Patch0: audiofile-CVE-2015-7747.patch
Patch1: audiofile-0.3.6-left-shift-neg.patch Patch1: audiofile-fix-gcc6-compile-error.patch
Patch2: audiofile-0.3.6-narrowing.patch
Patch3: audiofile-0.3.6-pull42.patch
Patch4: audiofile-0.3.6-pull43.patch
Patch5: audiofile-0.3.6-pull44.patch
BuildRequires: gcc-c++ libtool alsa-lib-devel flac-devel BuildRequires: gcc-c++ libtool alsa-lib-devel flac-devel
@ -78,6 +74,12 @@ make check
%{_mandir}/man3/* %{_mandir}/man3/*
%changelog %changelog
* Thu Jan 9 2020 JeanLeo<liujianliu.liu@huawei.com> - 0.3.6-23
- Type:bugfix
- ID:NA
- SUG:NA
- DESC:update software package
* Fri Sep 27 2019 chengquan<chengquan3@huawei.com> - 0.3.6-22 * Fri Sep 27 2019 chengquan<chengquan3@huawei.com> - 0.3.6-22
- Type:bugfix - Type:bugfix
- ID:NA - ID:NA